where y(n) is Output, x(n) is Input, x(n-1) is a delayed copy of the input,
and y(n-1) is a delayed copy of the output.

This filter is a recursive IIR or Infinite Impulse Response filter.
It can be unstable depending on the values of the coefficients.

A thorough description of the digital filter theory needed to fully describe this filter is beyond the scope of this
document.
Calculating coefficients is non-intuitive; the interested user is referred to one of the standard texts on filter theory
(e.g., Moore, "Elements of Computer Music", section 2.4).

Special thanks to Robert Bristow-Johnson for contributing his filter equations to the
music-dsp list.
They were used for calculating the coefficients for the lowPass, highPass, and other parametric filter calculations.